Direct SIP Feature Not Working After Upgrade to v20

[jawi]

Hi everyone,

We’ve been using the Direct SIP feature under version 18, and it worked flawlessly for our needs. However, after upgrading to version 20, the feature no longer functions properly.

Here’s what we’ve done so far:

• The Direct SIP option is enabled in v20.
• The SIP ID is correctly configured for the extension.
• Calls do come through, but the phone does not ring. Instead, after the calling party hangs up, we immediately receive a missed call notification.
• We are using the default port 5060 and our hosted address is like xyz.my3cx.de.
• The SRV records are correctly set, but since this is a hosted environment, we cannot modify them.

Has anyone encountered similar behavior or found a solution? Any help or insights would be greatly appreciated!

Thanks in advance!

Best regards,
Jan

 

[conversant]

We have encountered exactly the same issue. We have also tried enabling "PBX delivers audio" hoping that might address the issue with no success. We have also noticed that the call doesn't follow any of the programmed routes, so it never goes to voicemail at the target extension, nor does it follow any changes in call forwarding if we attempt to redirect the call to another extension. We are still working the issue. Have you had any success getting it working?

 

[KyriacosS_3CX]

Hi jawi,

Do you have any debug logs of such a call coming in?

Do the extensions have their direct SIP IDs set?

 

[paulodagraca]

Hello jawi,

What you do describe is an external call (from WAN/internet).
Did you try an internal call to the SIP-ID, does that work fine?

Meaning you can try to find out if the problem is the SIP-ID in general, or just the external (WAN/FQDN) that has this problem.
And do you use a name or just a number?

Paulo